Can you post the .call file (with called number blacked out) before call and after 1-2 calls? (file 1 should be before you mv to /v/s/a/o, file 2 should be from /v/s/a/o).
-----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brandon Phelps Sent: Monday, August 29, 2011 8:45 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Possible Bug? .call files executing multiple times On 08/19/2011 09:14 AM, Brandon Phelps wrote: > Hello all, > > We are setting up an auto-dialer to call customers based on the > opening of tickets in our internal ticketing system. Everything is > going fine so far except for one snag: > > To test the system we are implementing I am manually moving .call > files into the /var/spool/asterisk/outgoing directory like this: > > asterisk@dialerdev:~# cp test5703.call /tmp/test.call && mv > /tmp/test.call /var/spool/asterisk/outgoing/ > > This works great and the call is immediately started, however more > often than not (ie. not all the time, but most of the time) after > answering the call or rejecting it (sending it to voicemail), another > call is performed using the same file. > > I notice that when a call is initiated the .call file is not removed > immediately. Instead, asterisk waits until the call is completed > before removing the call file, so it seems like 5-10 seconds into the > call since the .call file still exists another call is placed. > > Any advice on how we can avoid this situation and ensure that only one > call is made per .call file? > > The OS is Ubuntu 11.04 server and we're running Asterisk 1.8. > > Thanks, > Sorry to bring this back up but I am still having this issue and haven't had any luck resolving it. It should be noted that the .call files in question are set to MaxRetries: 0, and simply connect the call to the 's' extension in a custom context. From there the context is pretty complicated, running some AGI scripts along with some dealing with user input, basically a simple IVR. Any help would be appreciated. Thanks, Brandon -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users